Security releases of Asterisk were recently created. In this post, we’d like to go into the depths of two of the security issues and how they affected Asterisk. Before going any further, I want to extend thanks to the following people for their help on this. Sandro Gauci (www.enablesecurity.com): Sandro was the person that reported
Continuous integration isn’t something that many people in the Asterisk project think about but it is a critical part of the development of Asterisk. It provides assurances on changes that go in and allows us to find problems faster, usually before a release occurs. The first introduction someone may have to our continuous integration is
Do you use WebRTC with Asterisk? Did you notice calls stop working after updating Google Chrome to version 57? Are you curious why that happened? The answer is the rtcp-mux feature. What is rtcp-mux? The majority of VoIP protocols make use of the Realtime Transmission Protocol (RTP) for transmitting and receiving media. In addition to
So you’ve heard there is now an Opus codec for Asterisk that’s been released. However, you are having problems with poor audio quality due to packets being dropped or lost. You’ve also heard or read that Opus can do something called FEC, but are not sure how do get it work with Asterisk. You’re in
There are several handler routines available to allow you to customize behavior for the different states of a call. Handler routines execute outside of the normal dialplan execution flow. It makes no sense to use the Hangup application in any of them and you must return from all of them. Most of the handlers operate
Warning, these tips may not be shocking, but you already know that and you clicked anyway! Shame! Many of us know the frustrating feeling of submitting a bug report and seeing it go through a triage process only to then watch it sit for hours, days, weeks with no response or updates. No fun at all! Yet
This week, we’re pleased to say that we’ve updated the Asterisk 13, 14 and master branches’ bundled version of pjproject to 2.6.
In the previous post, Josh introduced the forthcoming addition of streams to Asterisk. I’m going to piggyback on that to introduce a unified SDP API to Asterisk. What’s the motivation behind these additions? In recent years, Asterisk has risen above its old branding as a PBX. Support for WebRTC, combined with a REST API, has
Media streams are something that we all use every day. From watching videos on Youtube to placing calls media streams are right there with us in the background. They are the flow of media between two entities. They may be bidirectional (like a phone call) or they may even be unidirectional (like the Youtube video).
The recently announced Opus codec for Asterisk exposes a few configuration options that allow you to manipulate the encoder for your particular setup. These options can be set within codecs.conf. They are useful for customizing a format type that can then be specified on the “allow” line of an endpoint. The encoder uses all the
